1/* 2 * AMR narrowband decoder 3 * Copyright (c) 2006-2007 Robert Swain 4 * Copyright (c) 2009 Colin McQuillan 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * FFmpeg is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with FFmpeg; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23 24/** 25 * @file 26 * AMR narrowband decoder 27 * 28 * This decoder uses floats for simplicity and so is not bit-exact. One 29 * difference is that differences in phase can accumulate. The test sequences 30 * in 3GPP TS 26.074 can still be useful. 31 * 32 * - Comparing this file's output to the output of the ref decoder gives a 33 * PSNR of 30 to 80. Plotting the output samples shows a difference in 34 * phase in some areas. 35 * 36 * - Comparing both decoders against their input, this decoder gives a similar 37 * PSNR. If the test sequence homing frames are removed (this decoder does 38 * not detect them), the PSNR is at least as good as the reference on 140 39 * out of 169 tests. 40 */ 41 42 43#include <string.h> 44#include <math.h> 45 46#include "libavutil/channel_layout.h" 47#include "libavutil/float_dsp.h" 48#include "avcodec.h" 49#include "libavutil/common.h" 50#include "libavutil/avassert.h" 51#include "celp_math.h" 52#include "celp_filters.h" 53#include "acelp_filters.h" 54#include "acelp_vectors.h" 55#include "acelp_pitch_delay.h" 56#include "lsp.h" 57#include "amr.h" 58#include "internal.h" 59 60#include "amrnbdata.h" 61 62#define AMR_BLOCK_SIZE 160 ///< samples per frame 63#define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow 64 65/** 66 * Scale from constructed speech to [-1,1] 67 * 68 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but 69 * upscales by two (section 6.2.2). 70 * 71 * Fundamentally, this scale is determined by energy_mean through 72 * the fixed vector contribution to the excitation vector. 73 */ 74#define AMR_SAMPLE_SCALE (2.0 / 32768.0) 75 76/** Prediction factor for 12.2kbit/s mode */ 77#define PRED_FAC_MODE_12k2 0.65 78 79#define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz 80#define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter 81#define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode 82 83/** Initial energy in dB. Also used for bad frames (unimplemented). */ 84#define MIN_ENERGY -14.0 85 86/** Maximum sharpening factor 87 * 88 * The specification says 0.8, which should be 13107, but the reference C code 89 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.) 90 */ 91#define SHARP_MAX 0.79449462890625 92 93/** Number of impulse response coefficients used for tilt factor */ 94#define AMR_TILT_RESPONSE 22 95/** Tilt factor = 1st reflection coefficient * gamma_t */ 96#define AMR_TILT_GAMMA_T 0.8 97/** Adaptive gain control factor used in post-filter */ 98#define AMR_AGC_ALPHA 0.9 99 100typedef struct AMRContext { 101 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) 102 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0 103 enum Mode cur_frame_mode; 104 105 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe 106 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame 107 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame 108 109 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing 110 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector 111 112 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes 113 114 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe 115 116 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history 117 float *excitation; ///< pointer to the current excitation vector in excitation_buf 118 119 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector 120 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames) 121 122 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes 123 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes 124 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes 125 126 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX] 127 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65 128 uint8_t hang_count; ///< the number of subframes since a hangover period started 129 130 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset" 131 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none 132 uint8_t ir_filter_onset; ///< flag for impulse response filter strength 133 134 float postfilter_mem[10]; ///< previous intermediate values in the formant filter 135 float tilt_mem; ///< previous input to tilt compensation filter 136 float postfilter_agc; ///< previous factor used for adaptive gain control 137 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter 138 139 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples 140 141 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs 142 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs 143 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs 144 CELPMContext celpm_ctx; ///< context for fixed point math operations 145 146} AMRContext; 147 148/** Double version of ff_weighted_vector_sumf() */ 149static void weighted_vector_sumd(double *out, const double *in_a, 150 const double *in_b, double weight_coeff_a, 151 double weight_coeff_b, int length) 152{ 153 int i; 154 155 for (i = 0; i < length; i++) 156 out[i] = weight_coeff_a * in_a[i] 157 + weight_coeff_b * in_b[i]; 158} 159 160static av_cold int amrnb_decode_init(AVCodecContext *avctx) 161{ 162 AMRContext *p = avctx->priv_data; 163 int i; 164 165 if (avctx->channels > 1) { 166 avpriv_report_missing_feature(avctx, "multi-channel AMR"); 167 return AVERROR_PATCHWELCOME; 168 } 169 170 avctx->channels = 1; 171 avctx->channel_layout = AV_CH_LAYOUT_MONO; 172 if (!avctx->sample_rate) 173 avctx->sample_rate = 8000; 174 avctx->sample_fmt = AV_SAMPLE_FMT_FLT; 175 176 // p->excitation always points to the same position in p->excitation_buf 177 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; 178 179 for (i = 0; i < LP_FILTER_ORDER; i++) { 180 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); 181 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); 182 } 183 184 for (i = 0; i < 4; i++) 185 p->prediction_error[i] = MIN_ENERGY; 186 187 ff_acelp_filter_init(&p->acelpf_ctx); 188 ff_acelp_vectors_init(&p->acelpv_ctx); 189 ff_celp_filter_init(&p->celpf_ctx); 190 ff_celp_math_init(&p->celpm_ctx); 191 192 return 0; 193} 194 195 196/** 197 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters. 198 * 199 * The order of speech bits is specified by 3GPP TS 26.101. 200 * 201 * @param p the context 202 * @param buf pointer to the input buffer 203 * @param buf_size size of the input buffer 204 * 205 * @return the frame mode 206 */ 207static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, 208 int buf_size) 209{ 210 enum Mode mode; 211 212 // Decode the first octet. 213 mode = buf[0] >> 3 & 0x0F; // frame type 214 p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit 215 216 if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) { 217 return NO_DATA; 218 } 219 220 if (mode < MODE_DTX) 221 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1, 222 amr_unpacking_bitmaps_per_mode[mode]); 223 224 return mode; 225} 226 227 228/// @name AMR pitch LPC coefficient decoding functions 229/// @{ 230 231/** 232 * Interpolate the LSF vector (used for fixed gain smoothing). 233 * The interpolation is done over all four subframes even in MODE_12k2. 234 * 235 * @param[in] ctx The Context 236 * @param[in,out] lsf_q LSFs in [0,1] for each subframe 237 * @param[in] lsf_new New LSFs in [0,1] for subframe 4 238 */ 239static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) 240{ 241 int i; 242 243 for (i = 0; i < 4; i++) 244 ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, 245 0.25 * (3 - i), 0.25 * (i + 1), 246 LP_FILTER_ORDER); 247} 248 249/** 250 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector. 251 * 252 * @param p the context 253 * @param lsp output LSP vector 254 * @param lsf_no_r LSF vector without the residual vector added 255 * @param lsf_quantizer pointers to LSF dictionary tables 256 * @param quantizer_offset offset in tables 257 * @param sign for the 3 dictionary table 258 * @param update store data for computing the next frame's LSFs 259 */ 260static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER], 261 const float lsf_no_r[LP_FILTER_ORDER], 262 const int16_t *lsf_quantizer[5], 263 const int quantizer_offset, 264 const int sign, const int update) 265{ 266 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector 267 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector 268 int i; 269 270 for (i = 0; i < LP_FILTER_ORDER >> 1; i++) 271 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset], 272 2 * sizeof(*lsf_r)); 273 274 if (sign) { 275 lsf_r[4] *= -1; 276 lsf_r[5] *= -1; 277 } 278 279 if (update) 280 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); 281 282 for (i = 0; i < LP_FILTER_ORDER; i++) 283 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0); 284 285 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); 286 287 if (update) 288 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); 289 290 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); 291} 292 293/** 294 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors. 295 * 296 * @param p pointer to the AMRContext 297 */ 298static void lsf2lsp_5(AMRContext *p) 299{ 300 const uint16_t *lsf_param = p->frame.lsf; 301 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector 302 const int16_t *lsf_quantizer[5]; 303 int i; 304 305 lsf_quantizer[0] = lsf_5_1[lsf_param[0]]; 306 lsf_quantizer[1] = lsf_5_2[lsf_param[1]]; 307 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1]; 308 lsf_quantizer[3] = lsf_5_4[lsf_param[3]]; 309 lsf_quantizer[4] = lsf_5_5[lsf_param[4]]; 310 311 for (i = 0; i < LP_FILTER_ORDER; i++) 312 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i]; 313 314 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0); 315 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1); 316 317 // interpolate LSP vectors at subframes 1 and 3 318 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER); 319 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER); 320} 321 322/** 323 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. 324 * 325 * @param p pointer to the AMRContext 326 */ 327static void lsf2lsp_3(AMRContext *p) 328{ 329 const uint16_t *lsf_param = p->frame.lsf; 330 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector 331 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector 332 const int16_t *lsf_quantizer; 333 int i, j; 334 335 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]]; 336 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r)); 337 338 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)]; 339 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r)); 340 341 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]]; 342 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r)); 343 344 // calculate mean-removed LSF vector and add mean 345 for (i = 0; i < LP_FILTER_ORDER; i++) 346 lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0); 347 348 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); 349 350 // store data for computing the next frame's LSFs 351 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); 352 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); 353 354 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER); 355 356 // interpolate LSP vectors at subframes 1, 2 and 3 357 for (i = 1; i <= 3; i++) 358 for(j = 0; j < LP_FILTER_ORDER; j++) 359 p->lsp[i-1][j] = p->prev_lsp_sub4[j] + 360 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i; 361} 362 363/// @} 364 365 366/// @name AMR pitch vector decoding functions 367/// @{ 368 369/** 370 * Like ff_decode_pitch_lag(), but with 1/6 resolution 371 */ 372static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index, 373 const int prev_lag_int, const int subframe) 374{ 375 if (subframe == 0 || subframe == 2) { 376 if (pitch_index < 463) { 377 *lag_int = (pitch_index + 107) * 10923 >> 16; 378 *lag_frac = pitch_index - *lag_int * 6 + 105; 379 } else { 380 *lag_int = pitch_index - 368; 381 *lag_frac = 0; 382 } 383 } else { 384 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1; 385 *lag_frac = pitch_index - *lag_int * 6 - 3; 386 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2, 387 PITCH_DELAY_MAX - 9); 388 } 389} 390 391static void decode_pitch_vector(AMRContext *p, 392 const AMRNBSubframe *amr_subframe, 393 const int subframe) 394{ 395 int pitch_lag_int, pitch_lag_frac; 396 enum Mode mode = p->cur_frame_mode; 397 398 if (p->cur_frame_mode == MODE_12k2) { 399 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac, 400 amr_subframe->p_lag, p->pitch_lag_int, 401 subframe); 402 } else 403 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac, 404 amr_subframe->p_lag, 405 p->pitch_lag_int, subframe, 406 mode != MODE_4k75 && mode != MODE_5k15, 407 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6)); 408 409 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t 410 411 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2); 412 413 pitch_lag_int += pitch_lag_frac > 0; 414 415 /* Calculate the pitch vector by interpolating the past excitation at the 416 pitch lag using a b60 hamming windowed sinc function. */ 417 p->acelpf_ctx.acelp_interpolatef(p->excitation, 418 p->excitation + 1 - pitch_lag_int, 419 ff_b60_sinc, 6, 420 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0), 421 10, AMR_SUBFRAME_SIZE); 422 423 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float)); 424} 425 426/// @} 427 428 429/// @name AMR algebraic code book (fixed) vector decoding functions 430/// @{ 431 432/** 433 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame. 434 */ 435static void decode_10bit_pulse(int code, int pulse_position[8], 436 int i1, int i2, int i3) 437{ 438 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of 439 // the 3 pulses and the upper 7 bits being coded in base 5 440 const uint8_t *positions = base_five_table[code >> 3]; 441 pulse_position[i1] = (positions[2] << 1) + ( code & 1); 442 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1); 443 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1); 444} 445 446/** 447 * Decode the algebraic codebook index to pulse positions and signs and 448 * construct the algebraic codebook vector for MODE_10k2. 449 * 450 * @param fixed_index positions of the eight pulses 451 * @param fixed_sparse pointer to the algebraic codebook vector 452 */ 453static void decode_8_pulses_31bits(const int16_t *fixed_index, 454 AMRFixed *fixed_sparse) 455{ 456 int pulse_position[8]; 457 int i, temp; 458 459 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1); 460 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5); 461 462 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of 463 // the 2 pulses and the upper 5 bits being coded in base 5 464 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5; 465 pulse_position[3] = temp % 5; 466 pulse_position[7] = temp / 5; 467 if (pulse_position[7] & 1) 468 pulse_position[3] = 4 - pulse_position[3]; 469 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1); 470 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1); 471 472 fixed_sparse->n = 8; 473 for (i = 0; i < 4; i++) { 474 const int pos1 = (pulse_position[i] << 2) + i; 475 const int pos2 = (pulse_position[i + 4] << 2) + i; 476 const float sign = fixed_index[i] ? -1.0 : 1.0; 477 fixed_sparse->x[i ] = pos1; 478 fixed_sparse->x[i + 4] = pos2; 479 fixed_sparse->y[i ] = sign; 480 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign; 481 } 482} 483 484/** 485 * Decode the algebraic codebook index to pulse positions and signs, 486 * then construct the algebraic codebook vector. 487 * 488 * nb of pulses | bits encoding pulses 489 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7 490 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9 491 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11 492 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13 493 * 494 * @param fixed_sparse pointer to the algebraic codebook vector 495 * @param pulses algebraic codebook indexes 496 * @param mode mode of the current frame 497 * @param subframe current subframe number 498 */ 499static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, 500 const enum Mode mode, const int subframe) 501{ 502 av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2); 503 504 if (mode == MODE_12k2) { 505 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); 506 } else if (mode == MODE_10k2) { 507 decode_8_pulses_31bits(pulses, fixed_sparse); 508 } else { 509 int *pulse_position = fixed_sparse->x; 510 int i, pulse_subset; 511 const int fixed_index = pulses[0]; 512 513 if (mode <= MODE_5k15) { 514 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1); 515 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset]; 516 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1]; 517 fixed_sparse->n = 2; 518 } else if (mode == MODE_5k9) { 519 pulse_subset = ((fixed_index & 1) << 1) + 1; 520 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset; 521 pulse_subset = (fixed_index >> 4) & 3; 522 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0); 523 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2; 524 } else if (mode == MODE_6k7) { 525 pulse_position[0] = (fixed_index & 7) * 5; 526 pulse_subset = (fixed_index >> 2) & 2; 527 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1; 528 pulse_subset = (fixed_index >> 6) & 2; 529 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2; 530 fixed_sparse->n = 3; 531 } else { // mode <= MODE_7k95 532 pulse_position[0] = gray_decode[ fixed_index & 7]; 533 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1; 534 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2; 535 pulse_subset = (fixed_index >> 9) & 1; 536 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3; 537 fixed_sparse->n = 4; 538 } 539 for (i = 0; i < fixed_sparse->n; i++) 540 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0; 541 } 542} 543 544/** 545 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2) 546 * 547 * @param p the context 548 * @param subframe unpacked amr subframe 549 * @param mode mode of the current frame 550 * @param fixed_sparse sparse respresentation of the fixed vector 551 */ 552static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode, 553 AMRFixed *fixed_sparse) 554{ 555 // The spec suggests the current pitch gain is always used, but in other 556 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2) 557 // so the codebook gain cannot depend on the quantized pitch gain. 558 if (mode == MODE_12k2) 559 p->beta = FFMIN(p->pitch_gain[4], 1.0); 560 561 fixed_sparse->pitch_lag = p->pitch_lag_int; 562 fixed_sparse->pitch_fac = p->beta; 563 564 // Save pitch sharpening factor for the next subframe 565 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from 566 // the fact that the gains for two subframes are jointly quantized. 567 if (mode != MODE_4k75 || subframe & 1) 568 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX); 569} 570/// @} 571 572 573/// @name AMR gain decoding functions 574/// @{ 575 576/** 577 * fixed gain smoothing 578 * Note that where the spec specifies the "spectrum in the q domain" 579 * in section 6.1.4, in fact frequencies should be used. 580 * 581 * @param p the context 582 * @param lsf LSFs for the current subframe, in the range [0,1] 583 * @param lsf_avg averaged LSFs 584 * @param mode mode of the current frame 585 * 586 * @return fixed gain smoothed 587 */ 588static float fixed_gain_smooth(AMRContext *p , const float *lsf, 589 const float *lsf_avg, const enum Mode mode) 590{ 591 float diff = 0.0; 592 int i; 593 594 for (i = 0; i < LP_FILTER_ORDER; i++) 595 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i]; 596 597 // If diff is large for ten subframes, disable smoothing for a 40-subframe 598 // hangover period. 599 p->diff_count++; 600 if (diff <= 0.65) 601 p->diff_count = 0; 602 603 if (p->diff_count > 10) { 604 p->hang_count = 0; 605 p->diff_count--; // don't let diff_count overflow 606 } 607 608 if (p->hang_count < 40) { 609 p->hang_count++; 610 } else if (mode < MODE_7k4 || mode == MODE_10k2) { 611 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0); 612 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + 613 p->fixed_gain[2] + p->fixed_gain[3] + 614 p->fixed_gain[4]) * 0.2; 615 return smoothing_factor * p->fixed_gain[4] + 616 (1.0 - smoothing_factor) * fixed_gain_mean; 617 } 618 return p->fixed_gain[4]; 619} 620 621/** 622 * Decode pitch gain and fixed gain factor (part of section 6.1.3). 623 * 624 * @param p the context 625 * @param amr_subframe unpacked amr subframe 626 * @param mode mode of the current frame 627 * @param subframe current subframe number 628 * @param fixed_gain_factor decoded gain correction factor 629 */ 630static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, 631 const enum Mode mode, const int subframe, 632 float *fixed_gain_factor) 633{ 634 if (mode == MODE_12k2 || mode == MODE_7k95) { 635 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ] 636 * (1.0 / 16384.0); 637 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain] 638 * (1.0 / 2048.0); 639 } else { 640 const uint16_t *gains; 641 642 if (mode >= MODE_6k7) { 643 gains = gains_high[amr_subframe->p_gain]; 644 } else if (mode >= MODE_5k15) { 645 gains = gains_low [amr_subframe->p_gain]; 646 } else { 647 // gain index is only coded in subframes 0,2 for MODE_4k75 648 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)]; 649 } 650 651 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0); 652 *fixed_gain_factor = gains[1] * (1.0 / 4096.0); 653 } 654} 655 656/// @} 657 658 659/// @name AMR preprocessing functions 660/// @{ 661 662/** 663 * Circularly convolve a sparse fixed vector with a phase dispersion impulse 664 * response filter (D.6.2 of G.729 and 6.1.5 of AMR). 665 * 666 * @param out vector with filter applied 667 * @param in source vector 668 * @param filter phase filter coefficients 669 * 670 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] } 671 */ 672static void apply_ir_filter(float *out, const AMRFixed *in, 673 const float *filter) 674{ 675 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2 676 filter2[AMR_SUBFRAME_SIZE]; 677 int lag = in->pitch_lag; 678 float fac = in->pitch_fac; 679 int i; 680 681 if (lag < AMR_SUBFRAME_SIZE) { 682 ff_celp_circ_addf(filter1, filter, filter, lag, fac, 683 AMR_SUBFRAME_SIZE); 684 685 if (lag < AMR_SUBFRAME_SIZE >> 1) 686 ff_celp_circ_addf(filter2, filter, filter1, lag, fac, 687 AMR_SUBFRAME_SIZE); 688 } 689 690 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE); 691 for (i = 0; i < in->n; i++) { 692 int x = in->x[i]; 693 float y = in->y[i]; 694 const float *filterp; 695 696 if (x >= AMR_SUBFRAME_SIZE - lag) { 697 filterp = filter; 698 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) { 699 filterp = filter1; 700 } else 701 filterp = filter2; 702 703 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE); 704 } 705} 706 707/** 708 * Reduce fixed vector sparseness by smoothing with one of three IR filters. 709 * Also know as "adaptive phase dispersion". 710 * 711 * This implements 3GPP TS 26.090 section 6.1(5). 712 * 713 * @param p the context 714 * @param fixed_sparse algebraic codebook vector 715 * @param fixed_vector unfiltered fixed vector 716 * @param fixed_gain smoothed gain 717 * @param out space for modified vector if necessary 718 */ 719static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse, 720 const float *fixed_vector, 721 float fixed_gain, float *out) 722{ 723 int ir_filter_nr; 724 725 if (p->pitch_gain[4] < 0.6) { 726 ir_filter_nr = 0; // strong filtering 727 } else if (p->pitch_gain[4] < 0.9) { 728 ir_filter_nr = 1; // medium filtering 729 } else 730 ir_filter_nr = 2; // no filtering 731 732 // detect 'onset' 733 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) { 734 p->ir_filter_onset = 2; 735 } else if (p->ir_filter_onset) 736 p->ir_filter_onset--; 737 738 if (!p->ir_filter_onset) { 739 int i, count = 0; 740 741 for (i = 0; i < 5; i++) 742 if (p->pitch_gain[i] < 0.6) 743 count++; 744 if (count > 2) 745 ir_filter_nr = 0; 746 747 if (ir_filter_nr > p->prev_ir_filter_nr + 1) 748 ir_filter_nr--; 749 } else if (ir_filter_nr < 2) 750 ir_filter_nr++; 751 752 // Disable filtering for very low level of fixed_gain. 753 // Note this step is not specified in the technical description but is in 754 // the reference source in the function Ph_disp. 755 if (fixed_gain < 5.0) 756 ir_filter_nr = 2; 757 758 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2 759 && ir_filter_nr < 2) { 760 apply_ir_filter(out, fixed_sparse, 761 (p->cur_frame_mode == MODE_7k95 ? 762 ir_filters_lookup_MODE_7k95 : 763 ir_filters_lookup)[ir_filter_nr]); 764 fixed_vector = out; 765 } 766 767 // update ir filter strength history 768 p->prev_ir_filter_nr = ir_filter_nr; 769 p->prev_sparse_fixed_gain = fixed_gain; 770 771 return fixed_vector; 772} 773 774/// @} 775 776 777/// @name AMR synthesis functions 778/// @{ 779 780/** 781 * Conduct 10th order linear predictive coding synthesis. 782 * 783 * @param p pointer to the AMRContext 784 * @param lpc pointer to the LPC coefficients 785 * @param fixed_gain fixed codebook gain for synthesis 786 * @param fixed_vector algebraic codebook vector 787 * @param samples pointer to the output speech samples 788 * @param overflow 16-bit overflow flag 789 */ 790static int synthesis(AMRContext *p, float *lpc, 791 float fixed_gain, const float *fixed_vector, 792 float *samples, uint8_t overflow) 793{ 794 int i; 795 float excitation[AMR_SUBFRAME_SIZE]; 796 797 // if an overflow has been detected, the pitch vector is scaled down by a 798 // factor of 4 799 if (overflow) 800 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 801 p->pitch_vector[i] *= 0.25; 802 803 p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, 804 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE); 805 806 // emphasize pitch vector contribution 807 if (p->pitch_gain[4] > 0.5 && !overflow) { 808 float energy = p->celpm_ctx.dot_productf(excitation, excitation, 809 AMR_SUBFRAME_SIZE); 810 float pitch_factor = 811 p->pitch_gain[4] * 812 (p->cur_frame_mode == MODE_12k2 ? 813 0.25 * FFMIN(p->pitch_gain[4], 1.0) : 814 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX)); 815 816 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 817 excitation[i] += pitch_factor * p->pitch_vector[i]; 818 819 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, 820 AMR_SUBFRAME_SIZE); 821 } 822 823 p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation, 824 AMR_SUBFRAME_SIZE, 825 LP_FILTER_ORDER); 826 827 // detect overflow 828 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 829 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) { 830 return 1; 831 } 832 833 return 0; 834} 835 836/// @} 837 838 839/// @name AMR update functions 840/// @{ 841 842/** 843 * Update buffers and history at the end of decoding a subframe. 844 * 845 * @param p pointer to the AMRContext 846 */ 847static void update_state(AMRContext *p) 848{ 849 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0])); 850 851 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE], 852 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float)); 853 854 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float)); 855 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float)); 856 857 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE], 858 LP_FILTER_ORDER * sizeof(float)); 859} 860 861/// @} 862 863 864/// @name AMR Postprocessing functions 865/// @{ 866 867/** 868 * Get the tilt factor of a formant filter from its transfer function 869 * 870 * @param p The Context 871 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator 872 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator 873 */ 874static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d) 875{ 876 float rh0, rh1; // autocorrelation at lag 0 and 1 877 878 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf 879 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 }; 880 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response 881 882 hf[0] = 1.0; 883 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); 884 p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf, 885 AMR_TILT_RESPONSE, 886 LP_FILTER_ORDER); 887 888 rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE); 889 rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1); 890 891 // The spec only specifies this check for 12.2 and 10.2 kbit/s 892 // modes. But in the ref source the tilt is always non-negative. 893 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; 894} 895 896/** 897 * Perform adaptive post-filtering to enhance the quality of the speech. 898 * See section 6.2.1. 899 * 900 * @param p pointer to the AMRContext 901 * @param lpc interpolated LP coefficients for this subframe 902 * @param buf_out output of the filter 903 */ 904static void postfilter(AMRContext *p, float *lpc, float *buf_out) 905{ 906 int i; 907 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input 908 909 float speech_gain = p->celpm_ctx.dot_productf(samples, samples, 910 AMR_SUBFRAME_SIZE); 911 912 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter 913 const float *gamma_n, *gamma_d; // Formant filter factor table 914 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients 915 916 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) { 917 gamma_n = ff_pow_0_7; 918 gamma_d = ff_pow_0_75; 919 } else { 920 gamma_n = ff_pow_0_55; 921 gamma_d = ff_pow_0_7; 922 } 923 924 for (i = 0; i < LP_FILTER_ORDER; i++) { 925 lpc_n[i] = lpc[i] * gamma_n[i]; 926 lpc_d[i] = lpc[i] * gamma_d[i]; 927 } 928 929 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); 930 p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, 931 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); 932 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, 933 sizeof(float) * LP_FILTER_ORDER); 934 935 p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n, 936 pole_out + LP_FILTER_ORDER, 937 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); 938 939 ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out, 940 AMR_SUBFRAME_SIZE); 941 942 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE, 943 AMR_AGC_ALPHA, &p->postfilter_agc); 944} 945 946/// @} 947 948static int amrnb_decode_frame(AVCodecContext *avctx, void *data, 949 int *got_frame_ptr, AVPacket *avpkt) 950{ 951 952 AMRContext *p = avctx->priv_data; // pointer to private data 953 AVFrame *frame = data; 954 const uint8_t *buf = avpkt->data; 955 int buf_size = avpkt->size; 956 float *buf_out; // pointer to the output data buffer 957 int i, subframe, ret; 958 float fixed_gain_factor; 959 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing 960 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing 961 float synth_fixed_gain; // the fixed gain that synthesis should use 962 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use 963 964 /* get output buffer */ 965 frame->nb_samples = AMR_BLOCK_SIZE; 966 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 967 return ret; 968 buf_out = (float *)frame->data[0]; 969 970 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); 971 if (p->cur_frame_mode == NO_DATA) { 972 av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n"); 973 return AVERROR_INVALIDDATA; 974 } 975 if (p->cur_frame_mode == MODE_DTX) { 976 avpriv_report_missing_feature(avctx, "dtx mode"); 977 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); 978 return AVERROR_PATCHWELCOME; 979 } 980 981 if (p->cur_frame_mode == MODE_12k2) { 982 lsf2lsp_5(p); 983 } else 984 lsf2lsp_3(p); 985 986 for (i = 0; i < 4; i++) 987 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); 988 989 for (subframe = 0; subframe < 4; subframe++) { 990 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; 991 992 decode_pitch_vector(p, amr_subframe, subframe); 993 994 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, 995 p->cur_frame_mode, subframe); 996 997 // The fixed gain (section 6.1.3) depends on the fixed vector 998 // (section 6.1.2), but the fixed vector calculation uses 999 // pitch sharpening based on the on the pitch gain (section 6.1.3). 1000 // So the correct order is: pitch gain, pitch sharpening, fixed gain. 1001 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, 1002 &fixed_gain_factor); 1003 1004 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); 1005 1006 if (fixed_sparse.pitch_lag == 0) { 1007 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n"); 1008 return AVERROR_INVALIDDATA; 1009 } 1010 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, 1011 AMR_SUBFRAME_SIZE); 1012 1013 p->fixed_gain[4] = 1014 ff_amr_set_fixed_gain(fixed_gain_factor, 1015 p->celpm_ctx.dot_productf(p->fixed_vector, 1016 p->fixed_vector, 1017 AMR_SUBFRAME_SIZE) / 1018 AMR_SUBFRAME_SIZE, 1019 p->prediction_error, 1020 energy_mean[p->cur_frame_mode], energy_pred_fac); 1021 1022 // The excitation feedback is calculated without any processing such 1023 // as fixed gain smoothing. This isn't mentioned in the specification. 1024 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 1025 p->excitation[i] *= p->pitch_gain[4]; 1026 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], 1027 AMR_SUBFRAME_SIZE); 1028 1029 // In the ref decoder, excitation is stored with no fractional bits. 1030 // This step prevents buzz in silent periods. The ref encoder can 1031 // emit long sequences with pitch factor greater than one. This 1032 // creates unwanted feedback if the excitation vector is nonzero. 1033 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) 1034 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 1035 p->excitation[i] = truncf(p->excitation[i]); 1036 1037 // Smooth fixed gain. 1038 // The specification is ambiguous, but in the reference source, the 1039 // smoothed value is NOT fed back into later fixed gain smoothing. 1040 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], 1041 p->lsf_avg, p->cur_frame_mode); 1042 1043 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, 1044 synth_fixed_gain, spare_vector); 1045 1046 if (synthesis(p, p->lpc[subframe], synth_fixed_gain, 1047 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) 1048 // overflow detected -> rerun synthesis scaling pitch vector down 1049 // by a factor of 4, skipping pitch vector contribution emphasis 1050 // and adaptive gain control 1051 synthesis(p, p->lpc[subframe], synth_fixed_gain, 1052 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); 1053 1054 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); 1055 1056 // update buffers and history 1057 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); 1058 update_state(p); 1059 } 1060 1061 p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, 1062 buf_out, highpass_zeros, 1063 highpass_poles, 1064 highpass_gain * AMR_SAMPLE_SCALE, 1065 p->high_pass_mem, AMR_BLOCK_SIZE); 1066 1067 /* Update averaged lsf vector (used for fixed gain smoothing). 1068 * 1069 * Note that lsf_avg should not incorporate the current frame's LSFs 1070 * for fixed_gain_smooth. 1071 * The specification has an incorrect formula: the reference decoder uses 1072 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ 1073 p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], 1074 0.84, 0.16, LP_FILTER_ORDER); 1075 1076 *got_frame_ptr = 1; 1077 1078 /* return the amount of bytes consumed if everything was OK */ 1079 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC 1080} 1081 1082 1083AVCodec ff_amrnb_decoder = { 1084 .name = "amrnb", 1085 .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"), 1086 .type = AVMEDIA_TYPE_AUDIO, 1087 .id = AV_CODEC_ID_AMR_NB, 1088 .priv_data_size = sizeof(AMRContext), 1089 .init = amrnb_decode_init, 1090 .decode = amrnb_decode_frame, 1091 .capabilities = CODEC_CAP_DR1, 1092 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, 1093 AV_SAMPLE_FMT_NONE }, 1094}; 1095